[OpenSIPS-Users] Opensips + asterisk 1.4

[Willian Mazzardo - SYSSVOIP]

Hi Stephens... how do I do this?

Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030


2013/7/17 Stephen Vigus <svigus at gmail.com>

> Hi Willian
>
> You most likely need to configure Asterisk to not authenticate SIP
> requests coming from Opensips.
>
> Regards
> Stephen
>
>
>
> On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP <
> willian at syssvoip.com.br> wrote:
>
>> Hi all..
>>
>> I know this is a very simple scenario, all PSTN calls be routed to
>> asterisk to do the billing job, but im having some problems, this is my
>> scenario:
>>
>> Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247) .....
>> > PSTN
>>
>> Calls between sip clients on Opensips are working, but when I try to call
>> over Asterisk, I have Proxy authentication problem.
>>
>> Here is my logs:
>>
>> Opensips: http://pastebin.com/SWpuRHku
>> Asterisk: http://pastebin.com/6jp50LSS
>>
>> [opensips]
>> host=10.1.1.2
>> type=friend
>> context=callingcard
>> qualify=no
>> insecure=very
>> fromdomain=10.1.1.2
>>
>>
>> Route: http://pastebin.com/mLgpXiNx
>>
>> Can someone help me on this?
>>
>> Thanks
>>
>>
>> Willian Mazzardo
>> Depto TI - SYSSVOIP
>> www.syssvoip.com.br
>> 55 3537 2030
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20130717/a533f36c/attachment.htm>